Thax Nikolay for pushing me to see log deeper. Let me describe the problem
in other way.
First, i dont use trunk. Two systems connected by this article
http://wiki.sipfoundry.org/display/xecsuserV4r0/sipXecs+to+sipXecs+Calling.
Second,I investigate, that I can call to cisco 7940 phone and sip pl
Tony,
I am attempting to get HA running on our existing sipX clustered system.
What we have now is a single Bridge to our ITSP, and 2 backend HA-enabled
SIPX Servers, 1 of which is a Redundant SIP Router, the other Primary.
The idea being that using DNS (as posted in various places) and SRV record
I said it must be a role of any HA system, even if your purpose is to run
sipxbridge on it and not create any users on it or purposely register users
to it.
I don't recall what you are trying to do (one server with role of sipxbridge
only) was a design intent (or supportable). What did you read or
Because you didn't word it well - or perhaps I misread - you're saying I
need to have SIP Proxy running on all backend servers?
If so that IS the case - on both SIPX1 and SIPX2 the SIP Proxy service (part
of the SIP Router role, both Primary and Redundant) is in fact running.
In response to Dave
On the Internet Calling->NAT Traversal page I have checked the box next
to "Server is behind NAT."
On the Gateways -> ISTP Account page SIP Keepalive is set to "None"
which is supposed to default to CR-LF and RTP keepalive is set to
"replay last packet."
Flowroute's account settings page doesn't
Can you confirm you system is behind nat, and your settings are as I
described?
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.9
With flowroute, your sip-keepalive should be CR-LF and the RTP keepalive
should be "replay last packet".
You can safely ignore their statement about NAT's networks, because
sipxbridge takes care of that for you. In ANY installation where sipx is
behind nat with a properly configured firewall YOU_S
I asked Flowroute what was their preferred method for SIP keepalive.
Apparently, they don't need it at all. Here's the reply:
The SIP keepalive is used by VoIP providers that cannot transverse NATed
networks, however, Flowroute'snetwork supports the use of NATed
systems as we are able
Thanks, Tony. I'll check with them.
On Mon 06.Sep.10 12:07, Tony Graziano wrote:
>You need to ask them what their preferred SIP KEEPALIVE method is, and
>change the gateway accordingly.
>
>Alternatively you can change the SIP KEEPALIVE method to a different value
>and wait 21 minutes to see if the
You need to ask them what their preferred SIP KEEPALIVE method is, and
change the gateway accordingly.
Alternatively you can change the SIP KEEPALIVE method to a different value
and wait 21 minutes to see if the calls till works. If you have an account
with them, they should be able to answer this
I've been having a problem with calls dropping. The ISTP (Flowroute)
sends a BYE after 20 minutes and the call ends. Is this more likely to
be a configuration problem on my end or a problem with the ISTP? Just
looking for pointers on where to start troubleshooting.
I'm running 4.2.1 (4.2.1-018971
A bit offtopic: regarding siptrunk between two sipx systems.
I think, that sometime it is convinient to use siptrunk.
The thing is that, when two sipx systems are "routable" but have different
"intranet subnets" one has to relay media via sipxbridge, albeit via internal
ip address.
Rgds,
Nikolay
Can you also explain how you configured your dialplan to call between the
two systems? A siptrunk would be unneeded in any version, so it would be
helpful to know what the dialplan looks like.
On Mon, Sep 6, 2010 at 6:13 AM, Nikolay Kondratyev wrote:
> Alexander,
> the info, you provided is not
Done.
http://track.sipfoundry.org/browse/XX-8848
Thanks and regards,
Nikolay.
_
From: Michael Picher [mailto:mpic...@gmail.com]
Sent: Thursday, September 02, 2010 3:07 PM
To: Nikolay Kondratyev
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Voicemailbox directories are not
Alexander,
the info, you provided is not enough.
The following, i guess, will be ok:
Set at least INFO log level for all the processes.
Provide two xml traces of the same failed call from both systems (see wiki
how to trace call using sipviewer, these traces will contain sipx internal
interprocess
I have 3.10.2 (10.0.15.2) and 4.2.1(10.0.20.2) sipxecs systems. Sip trunk
configured and works well. If I configure any caller ID for the extension
(or for the trunking gateway) on 4.2.1 system I can not call to 3.10.2
system.
Let's see 4.2.1 trace.
(1) is usual INVITE from 9011 extension (4.2
>From a network perspective 192.168.1.1/24 and 192.168.1.0/24 are both
describing the same network, the part with 0's in the subnetmask doesn't
matter.
What network would 192.168.1.1/24 mean otherwise?
So it should have worked if you ask me.
A checker to validate that the host part of the subnet
Interesting idea.
Maybe it makes sence to make the keys that are sent to active window
configurable...
I will see what I can do.
René
2010/9/3 Paul Scheepens
> ctrl-Insert might be more linux fiendly???
>
>
> Paul
> >
> > Of course it might be possible to send "ctrl&c" to the active window
> >
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